Hi Sebastian,
I will put together a small setup and see what is going wrong.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 17.08.2015 13:40, Sebastian Sastre wrote:
Bodgan,
Thanks i wasn't sure on the ack process. This is the log , the
scenario is triggered by a httpd json call.
INFO:b2b_logic:b2bl_add_client: adding entity
[0x7f718dfa7068]->[B2B.173.7331923] to tuple [0x7f718dfa0cd0]->[685.0]
WARNING:b2b_logic:b2bl_delete_entity: entity [0x7f718dfa2d18]->[] not
found for tuple [685.0]
INFO:b2b_logic:b2bl_delete_entity: delete tuple [685.0], entity []
INFO:b2b_logic:b2bl_add_client: adding entity
[0x7f718dfa4d28]->[B2B.173.5533781] to tuple [0x7f718dfa0cd0]->[685.0]
INFO:b2b_logic:b2b_add_dlginfo: Dialog pair: [B2B.173.7331923] -
[B2B.173.5533781]
and the trace looks like this
172.10.1.21 -> 172.10.1.107 SIP 436 Request: INVITE
sip:sebas3@172.10.1.107:5060 <http://sip:sebas3@172.10.1.107:5060>
172.10.1.107 -> 172.10.1.21 SIP 346 Status: 100 Giving a try
172.10.1.107 -> 172.10.1.21 SIP 456 Status: 180 Ringing
172.10.1.107 -> 172.10.1.21 SIP/SDP 1088 Status: 200 Ok, with session
description
172.10.1.21 -> 172.10.1.20 SIP/SDP 843 Request: INVITE
sip:1@172.10.1.20:5060 <http://sip:1@172.10.1.20:5060>, with session
description
172.10.1.20 -> 172.10.1.21 SIP 390 Status: 100 Trying
172.10.1.20 -> 172.10.1.21 SIP/SDP 1252 Status: 200 OK, with session
description
172.10.1.21 -> 172.10.1.107 SIP 526 Request: ACK
sip:sebas3@73.139.116.217 <mailto:sip%3Asebas3@73.139.116.217>
172.10.1.21 -> 172.10.1.20 SIP 446 Request: ACK
sip:1@172.10.1.20:5060;transport=udp
172.10.1.107 -> 172.10.1.21 SIP 474 Request: BYE
sip:DialerProxy@172.10.1.21:5060 <http://sip:DialerProxy@172.10.1.21:5060>
172.10.1.21 -> 172.10.1.20 SIP 446 Request: BYE
sip:1@172.10.1.20:5060;transport=udp
172.10.1.20 -> 172.10.1.21 SIP 545 Status: 200 OK
172.10.1.21 -> 172.10.1.107 SIP 629 Status: 200 OK
thanks !
On Mon, Aug 17, 2015 at 5:47 AM, Bogdan-Andrei Iancu
<bog...@opensips.org <mailto:bog...@opensips.org>> wrote:
Hi Sebastian,
The 200OK from FS must be followed by ACK+SDP to linphone. See:
http://www.opensips.org/Documentation/Tutorials-B2BUA#toc14
If this does not happen -> do you see any errors in the logs
(around the processing of 200OK from FS) ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 17.08.2015 04:18, Sebastian Sastre wrote:
Hi guys,
Im using the B2BUA module to send a call out to our subscribers
and bridge them with our IVR server on answer.
The subscriber side uses linphone and the media server is a
freeswitch 1.6. When placing the call thru the trigger scenario
MI command, the initial invite does not have any SDP inside which
makes sense.
Once the 200ok is received from the linphone client, opensips
uses the SDP contained in the 200 to generate an invite to the
freeswitch box. which is great.
However, when the 200ok is received from freeswitch, the
following ACK back the linphone client does not contain the SDP
and Linphone complains with "No codec intersection" and sends an
immediate bye.
Am i right to think that the sdp should go in the ack to create a
late offer?
Should i be sending a re invite?
any help appreciated.
My scenario is simple.
<?xml version="1.0"?>
<scenario id="dialer" name="MS start conditional" param="2"
type="extern">
<init>
<bridge>
<client>
<id>client1</id>
<destination>
<value type="param">1</value>
</destination>
</client>
<client>
<id>client2</id>
<destination>
<value type="param">2</value>
</destination>
</client>
</bridge>
<state>1</state>
</init>
</scenario>
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